System for compression and decompression of audio signals for digital transmision

ABSTRACT

A system for recognizing the existence of and adjusting the psycho-acoustic parameters present in an audio digital CODEC. A audio digital CODEC is provided with various parameters that when changed affect the quality of the resultant audio. These psycho-acoustic parameters include the standard ISO parameters and additional parameters to aid in effecting a pure resulting audio quality. The psycho-acoustic parameters located in the audio digital CODEC can be monitored and controlled by the user. The parameters can be monitored by a speaker associated with the CODEC or headphones. The user can control the adjustment of the psycho-acoustic parameters through the use of knobs present on the front panel of the CODEC or graphic or digital representations. Adjustment of the parameters will provide real time change of the resulting audio sound that the user can monitor through the speaker or the headphones.  
     DPPA permits the user to dynamically change the values of different parameters. The ability to change the parameters can be embodied in front panel knobs or in the action of computer software as instructed by the user.

RELATED APPLICATION

[0001] This application is a continuation of application Ser. No.08/420,721, Filed Apr. 10. 1995, entitled “System for AdjustablePsycho-Acoustic Parameters In A Digital Audio CODEC” and a continuationof application Ser. No. 08/419,200 filed Apr. 10, 1995. entitled “SystemFor Compression and Decompression of Audio For Digital Transmission”.

SOURCE CODE APPENDIX

[0002] Source code for the control processor of the present inventionhas been included as a microfiche SOURCE CODE APPENDIX.

FIELD OF THE INVENTION

[0003] The present invention relates generally to an audio CODEC for thecompression and decompression of audio signals for transmission overdigital facilities, and more specifically, relates to an audio CODECthat is programmable by a user to control various CODEC operations, suchas monitoring and adjusting a set of psycho-acoustic parameters,selecting, different modes of digital transmission, and downloading newcompression algorithms.

BACKGROUND OF THE INVENTION

[0004] Current technology permits the translation of analog audiosignals into a sequence of binary numbers (digital). These numbers maythen be transmitted and received through a variety of means. Thereceived signals may then be converted back into analog audio signals.The device for performing both the conversion from analog to digital andthe conversion from digital to analog is called a CODEC. This is anacronym for COder/DECoder.

[0005] The cost of transmitting bits from one location to another is afunction of the number of bits transmitted per second. The higher thebit transfer rate the higher the cost. Certain laws of physics in humanand audio perception establish a direct relationship between perceivedaudio quality and the number of bits transferred per second. The netresult is that improved audio quality increases the cost oftransmission.

[0006] CODEC manufacturers have developed technologies to reduce thenumber of bits required to transmit any given audio signal (compressiontechniques) thereby reducing the associated transmission costs. The costof transmitting bits is also a function of the transmission facilityused, i.e., satellite, PCM phone lines, ISDN (fiber optics).

[0007] A CODEC that contains some of these compression techniques alsoacts as a computing device. It inputs the analog audio signal, convertsthe audio signal to a digital bit stream, and then applies a compressiontechnique to the bit stream thereby reducing the number of bits requiredto successfully transmit the original audio signal. The receiving CODECapplies the same compression techniques in reverse (decompression) sothat it is able to convert the compressed digital bit stream back intoan analog audio signal. The difference in quality between the analogaudio input and the reconstructed audio output is an indication of thequality of the compression technique. The highest quality techniquewould yield an identical signal reconstruction.

[0008] Currently, the most successful compression techniques are calledperceptual coding techniques. These types of compression techniquesattempt to model the human ear. These compression techniques are basedon the recognition that much of what is given to the human ear isdiscarded because of the characteristics of the ear. For example, if aloud sound is presented to a human ear along with a softer sound, theear will only hear the loud sound. As a result, encoding compressiontechniques can effectively ignore the softer sound and not assign anybits to its transmission and reproduction under the assumption that ahuman listener can not hear the softer sound even if it is faithfullytransmitted and reproduced.

[0009] Many conventional CODECs use perceptual coding techniques whichutilize a basic set of parameters which determine their behavior. Forexample, the coding technique must determine how soft a sound must berelative to a louder sound in order to make the softer sound a candidatefor exclusion from transmission. A number which determines thisthreshold is considered a parameter of the scheme which is based on thatthreshold. These parameters are largely based on the human psychology ofperception so then are collectively known as psycho-acoustic parameters.

[0010] However, conventional CODECs which use perceptual coding haveexperienced limitations, more specifically, manufacturers of existingCODECs preprogram all of the CODECs operating variables which controlthe compression technique, decompression technique, bit allocation andtransmission rate. By preprogramming the CODEC, the manufacturerundesirably limits the user interaction with the resulting CODEC. Forexample, it is known that audio can be transmitted by digitaltransmission facilities. These digital transmissions include digitaldata services, such as conventional phone lines, ISDN, T1, and E1. Otherdigital transmission paths include RF transmission facilities such asspread spectrum RF transmission and satellite links.

[0011] Although existing CODECs can transmit compressed audio signalsvia digital transmission facilities, any variables regarding the mode oftransmission are preprogrammed by the manufacturer of the CODEC, therebylimiting the CODECs use to a single specific transmission facility.Hence, the user must select a CODEC which is preprogrammed to becompatible with the user's transmission facility. Moreover, existingCODECs operate based on inflexible compression and bit allocationtechniques and thus, do not provide users with a method or apparatus tomonitor or adjust the CODEC to fit the particular user's wants andneeds. Accordingly, users must test CODECs with different compressionand bit allocation techniques and then select the one device which hasthe features or options so desired, e.g. satellite transmissioncapabilities.

[0012] Moreover, standard coding techniques have been developed in orderto ensure interoperability of CODECs from different manufacturers and toensure an overall level of audio quality, thereby limiting the CODEC'suse to a single specific transmission facility. One such standard is theso-called ISO/MPEG Layer-II compression standard, for the compressionand decompression of an audio input. This standard sets forth acompression technique and a bit stream syntax for the transmission ofcompressed binary data. The ISO/MPEG Layer-II standard defines a set ofpsycho-acoustic parameters that is useful in performing compression.U.S. Pat. No. 4,972,484, entitled “Method of Transmitting or StoringMasked Sub-band Coded Audio Signals,” discloses the ISO/MPEG Layer-IIstandard and is incorporated by reference.

[0013] However, conventional CODECs do not use a uniform set ofparameters. Each CODEC manufacturer determines their own set ofpsycho-acoustic parameters either from a known standard or as modifiedby the manufacturer in an attempt to provide the highest quality soundwhile using the lowest number of bits to encode audio. Once themanufacturer selects a desired parameter set, the manufacturer programsvalues for each of the parameters. These preprogrammed parameter valuescorrespond to the manufacturer's perception of an optimal audio qualityat the decoder.

[0014] However, in conventional CODECs, users typically are unaware ofthe existence or nature of these parameters. Further, the user has nocontrol over the parameter values. As a result, users were required totest different CODECs from different manufacturers and then select theCODEC that met the user's requirements or that sounded best to the user.

[0015] Typically, conventional CODECs utilize standard parameters whichhave been accepted by the International Standards Organization (ISO) andhave been adopted as part of the International Standards Organization,Motion Picture Experts Group (ISO/MPEG) Layer-II compression standard.However, the ISO/MPEG Layer-II standard has met with limited acceptancesince these parameters do not necessarily provide CD quality output. TheISO/MPEG Layer-II parameters are determined and set based on the averagehuman ear. The parameters do not account for the variations between eachindividuals hearing capabilities. Hence, the conventional standards andCODECs do not afford the ability for users to tune their CODEC to theuser's individual subjective hearing criteria. Nor are conventionalCODECs able to meet changing audio needs and to shape the overall soundof their application.

[0016] A need remains within the industry for an improved CODEC which ismore flexible, programmable by the user, and which overcomes thedisadvantages experienced heretofore. It is an object of the presentinvention to meet this need.

OBJECTS OF THE INVENTION

[0017] It is an object of the present invention to provide aprogrammable audio CODEC that can be monitored, controlled and adjustedby a user to control the various functions of the CODEC.

[0018] It is another object of the present invention to provide an audioCODEC that is programmable by a user to transmit compressed digital bitstreams over various user selected digital transmission facilities.

[0019] It is an object of the present invention to provide a userprogrammable audio CODEC With a plurality of psycho-acoustic parametersthat can be monitored, controlled, and adjusted by a user to change theaudio output from the CODEC.

[0020] It is a related object of the present invention to provide anaudio CODEC with new psycho-acoustic parameters.

[0021] It is a further related object of the present invention toprovide an audio CODEC where the psycho-acoustic parameters are changedby knobs on the front panel of the CODEC.

[0022] It is another related object of the present invention to providean audio CODEC where the psycho-acoustic parameters are changed by akeypad on the front panel of the CODEC.

[0023] It is still a further related object of the present invention toprovide an audio CODEC with a personal computer connected thereto toadjust the psycho-acoustic parameters by changing graphicrepresentations of the parameters on a computer screen.

[0024] It is a related object of the present invention to provide anaudio CODEC that is programmable by a user to transmit compresseddigital bit streams over a digital data service.

[0025] It is a further related object of the present invention toprovide an audio CODEC that is programmable by a user for transmissionof compressed digital bit streams over any of T1, E1 and ISDN lines orover RF transmission facilities.

[0026] It is yet another related object of the present invention toprovide an audio CODEC that is user programmable for transmission ofcompressed digital bit streams via satellite.

[0027] It is a further object of the present invention to provide anaudio CODEC for transmission of asynchronous data together with thetransmission of compressed audio.

[0028] It is still a further object of the present invention to providean audio CODEC that utilizes the multiple audio compression anddecompression schemes.

[0029] It is still another object of the present invention to provide anaudio CODEC which allows a user to select one of several stored audiocompression techniques.

[0030] It is still another object of the present invention to provide anaudio CODEC that is remotely controlled by a host computer.

[0031] It is stilt another object of the present invention to provide anaudio CODEC for monitoring either the encoder input signal or thedecoder output signal with the use of headphones.

[0032] It is still another object of the present invention to provide anaudio CODEC with safeguards for automatically selecting a secondtransmission facility if a first user selected transmission facilityfails.

[0033] It is yet another object of the present invention to provide anaudio CODEC that can be controlled by inputting control commands into akey pad on the front panel of the CODEC.

[0034] It is related object of the present invention to provide an audioCODEC having a user interface to control and program the audio CODECthrough the use of a graphics display on the front panel.

[0035] It is still another related object of the present invention toprovide for connection of a personal computer to the audio CODEC forcontrolling the input of program information thereto.

[0036] It is still another object of the present invention to providebi-directional communication between two audio CODECs.

[0037] It is still another object of the present invention to provide anaudio CODEC that can be interfaced to a local area network.

[0038] It is yet another object of the present invention to provide anaudio CODEC that will provide programmed information to users throughthe use of indicators on the front panel of the CODEC.

[0039] It is yet another object of the present invention to provide anaudio CODEC that can send non-audio compressed information includingtext video and graphic information.

[0040] It is still another object of the present invention to provide anaudio CODEC that can store and retrieve information on and from anelectronic storage medium or a disk

[0041] It is still another related object of the present invention toprovide an audio CODEC that can transmit control information along withthe textual video and graphic information.

[0042] It is still a further object of the present invention to providedigital audio compression techniques that yield improved and preferablyCD quality audio.

[0043] It is a related object of the present invention to provide acompression scheme that yields better audio quality than the MPEGcompression standard.

[0044] It is still another related object of the present invention toprovide CD quality audio that achieves a 12 to 1 compression ratio.

SUMMARY OF THE INVENTION

[0045] The present invention provides a CODEC which holds severalcompression algorithms and allows the user easily to download futureaudio compression algorithms as needed. This makes the present CODECvery versatile and prevents it from becoming obsolete.

[0046] The preferred CODEC provides for both digital and analog input ofexternal signals. The CODEC is also capable of handling a wide varietyof ancillary data which can be incorporated into the compressed bitstream along with the audio and header data. The ancillary bit streampreferably enters the encoder directly from external sources. However,the user could alternatively choose to have the external datamultiplexed into a composite ancillary bit stream before being encodedwith the audio and header data. The preferred CODEC also provides forrate adaptation of signals that are input (and output) at one rate andcompressed (and decompressed) at yet another rate. This rate adaptationcan also be synchronized to external clock sources.

[0047] The user can also programmably alter the psycho-acousticcompression parameters to optimize transmissions under differentconditions. The disclosed invention also allows the user to programmablycontrol CODEC transmission modes as well as other CODEC operations. Suchprogrammable control is achieved through remote interfaces and/or directkeypad control.

[0048] The compressed output signal can also be interfaced with avariety of external sources through different types of output DigitalInterface Modules (DIMs). Similar input DIMs would input return signalsfor decoding and decompression by the CODEC. Certain specialized DIMsmight also operate as satellite receiver modules. Such modules wouldpreferably store digital information as it becomes available for laterediting and use. Satellite receiver modules would be capable ofreceiving information such as audio, video, text, and graphics. Thisinformation would then be decoded and decompressed as appropriate by theCODEC.

[0049] Additional features and advantages of the present invention willbecome apparent to one of skilled in the art upon consideration of thefollowing detailed description of the present invention.

BRIEF DESCRIPTIONS OF THE DRAWINGS

[0050]FIG. 1 is a block diagram of a CODEC illustrating signalconnections between various components in accordance with a preferredembodiment of the present invention.

[0051]FIG. 2 is a block diagram of a CODEC illustrating signalconnections between various components in accordance with the preferredembodiment shown in FIG. 1.

[0052]FIG. 3 is a block diagram illustrating ancillary data beingmultiplexed into a composite bit stream in accordance with the preferredembodiment of FIG. 1.

[0053]FIG. 4 is a block diagram illustrating an ISO/MPEG audio bitstream being decoded into a composite ancillary bit stream and audioleft and right signals in accordance with the preferred embodiment ofFIG. 1.

[0054]FIG. 5 is an example of a front panel user keypad layout inaccordance with a preferred embodiment of the present invention.

[0055]FIG. 6 is another example of a front panel user keypad layout inaccordance with a preferred embodiment of the present invention.

[0056]FIG. 7 is another example of a front panel user keypad layout inaccordance with a preferred embodiment of the present invention.

[0057]FIG. 8 is a block diagram showing the decoder output timing withthe AES/EBU sync disabled or not present and using normal timing inaccordance with a preferred embodiment of the present invention.

[0058]FIG. 9 is a block diagram showing the decoder output timing withAES/EBU sync disabled or not present using internal crystal timing inaccordance with a preferred embodiment of the present invention.

[0059]FIG. 10 is a block diagram showing decoder output timing withAES/EBU sync enabled and present using AES timing in accordance with apreferred embodiment of the present invention.

[0060]FIG. 11 is an example of an LED front panel display in accordancewith a preferred embodiment of the present invention.

[0061]FIG. 12 is another example of an LED front panel display inaccordance with a preferred embodiment of the present invention.

[0062]FIG. 13 is a block diagram of a CODEC illustrating signalconnections between various components allowing transmission of audio,video, text, and graphical information in accordance with a preferredembodiment of the present invention.

[0063]FIG. 14 is a diagram illustrating the interconnection betweenvarious modules in accordance with a preferred embodiment.

[0064]FIG. 15 is a block diagram of an embodiment of an encoder asimplemented in the CODEC of the system in accordance with the preferredembodiment shows in FIG. 14.

[0065]FIG. 16 is a diagram illustrating a known representation of atonal masker as received and recognized by a CODEC system.

[0066]FIG. 17 is a diagram illustrating a known representation of atonal masker and its associated masking skirts as recognized by a CODECsystem.

[0067] Figure is a diagram illustrating a tonal masker and itsassociated masking skirts as implemented by the encoder of the system inaccordance with the preferred embodiment shown in FIG. 14.

[0068]FIG. 19 is a diagram illustrating the representation of theaddition of two tonal maskers as implemented by the encoder of thesystem in accordance with the preferred embodiment shown in FIG. 14.

[0069]FIG. 20 is a block diagram illustrating the adjustment of a singleparameter as performed by the encoder of the system in accordance withthe preferred embodiment shown in FIG. 14.

[0070]FIG. 21 illustrates a block diagram of an encoder for a singleaudio channel according to the present invention.

[0071]FIG. 22 illustrates a data structure used in the preferredembodiment for a frame of data.

[0072]FIG. 23 illustrates a block diagram of an encoder for two audiochannels operated in joint stereo according to the present invention.

[0073]FIG. 24 illustrates a flow diagram of the process followed by thepresent invention when adjusting the scaling factors.

[0074]FIGS. 25a and 25 b illustrate a flow diagram of the overallprocess followed by the present invention when assigning encoding levelsto the quantizers.

[0075]FIG. 26 illustrates a flow diagram of the process followed by thepresent invention when obtaining a global masking threshold.

[0076]FIG. 27 illustrates a flow diagram of the process followed by thepresent invention predicting bit allocation for mono, stereo or jointstereo frames.

[0077]FIG. 28 illustrates a flow diagram of the process followed by thepresent invention when determining an allocation step for a specificsubband.

[0078]FIG. 29 illustrates a flow diagram of the process followed by thepresent invention when determining the joint stereo boundary.

[0079]FIG. 30 illustrates a flow diagram of the process followed by thepresent invention when assigning a quantization level.

[0080]FIG. 31 illustrates a flow diagram of the process followed by thepresent invention when deallocating bits from one or more subbandsfollowing the initial allocation process.

[0081]FIGS. 32a and 32 b illustrate graphs of exemplary subbands havinga portion of the global masking threshold therein and multiplemasking-to-noise ratios therein corresponding to multiple allocationsteps.

[0082]FIG. 33 illustrates a deallocation table recorded during bitallocation and deallocation.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

[0083] CODEC System with Adjustable Parameters

[0084] With reference to FIGS. 14 and 15, a CODEC 1010 has an encoder1012 and a decoder 1010. The encoder 1012 receives as input an analogaudio source 1016. The analog audio source 1016 is converted by ananalog to digital converter 1018 to a digital audio bit stream 1020. Theanalog to digital converter 1018 can be located before the encoder 1012,but is preferably contained therein. In the encoder 1012, compressiontechniques compress the digital audio bit stream 1020 to filter outunnecessary and redundant noises. In the preferred embodiment, thecompression technique utilizes the parameters defined by the ISO/MPEGLayer-II standard as described in U.S. Pat. No. 4,972,484, and in adocument entitled, “Information Technology Generic Coding Of MovingPictures And Associated Audio,” and is identified by citation ISO3-11172 Rev. 2. The '484 patent and the ISO 3-11172, Rev. 2 Document areincorporated by reference.

[0085] In addition, the compression technique of the preferredembodiment of the encoder 1012 adds several new parameters as explainedbelow. The resultant compressed digital audio bit stream 1022 is thentransmitted by various transmission facilities (not shown) to a decoderat another CODEC (not shown). The decoder decompresses the digital audiobit stream and then the digital bit stream is converted to an analogsignal.

[0086] The compression technique utilized by the CODEC 1010 to compressthe digital audio bit stream 1020 is attached as the Source CodeAppendix, and is hereby incorporated by reference.

[0087] Human Auditory Perception—Generally

[0088] The audio compression routine performed by the encoder 1012 ispremised on several phenomena of human auditory perception. While thosephenomena are generally understood and explained in the ISO Document and'484 patent referenced above, a brief summary is provided hereafter.

[0089] Generally, it is understood that when a human ear receives a loudsound and a soft sound, close in time, the human will only perceive theloud sound. In such a case, the loud sound is viewed as “masking” orcovering up the quiet or soft sound.

[0090] The degree to which the softer sound is masked is dependent, inpart, upon the frequencies of the loud and soft sounds and the distancebetween the frequencies of the loud and soft sounds. For instance, aloud sound at 700 Hz will have a greater masking effect upon a softsound at 750 Hz than upon a soft sound at 900 Hz. Further, typically,the ear is more discriminating between loud and soft sounds at lowfrequencies as compared to loud and soft sounds at high frequencies.

[0091] Another aspect of hearing and psycho-acoustics is that a personcan hear two tones at the same frequency provided that the softer toneis close enough in amplitude to the louder tone. The maximum differencein amplitude between the two tones of common frequency is referred to asthe masking index. The masking index is dependent, in part, uponfrequency of the tones. Generally, the masking index increases withfrequency. For instance, the masking index of a masking tone at 1000 Hzwill be smaller than the masking index of a masking tone at 7000 Hz.

[0092]FIG. 17 illustrates the masking index 1034 for the tonal masker1024. Thus, the masking effect will be greater for a loud sound at 7000Hz upon a soft sound 7050 Hz as compared to the masking effect of a loudsound at 700 Hz upon a soft sound at 750 Hz. The masking effect of asound is defined by its “masking skirt,” which is explained below.

[0093] The encoder defines maskers and masking skirts based on the abovenoted masking effects (as explained below in more detail). If maskingdoes occur, then the compression technique will filter out the masked(redundant) sound.

[0094] The audio compression technique of the encoder is also premisedon the assumption that there are two kinds of sound maskers. These twotypes of sound maskers are known as tonal and noise maskers. A tonalmasker will arise from audio signals that generate nearly pure,harmonically rich tones or signals. A tonal masker that is pure(extremely clear),will have a narrow bandwidth. The band width of atonal masker varies with frequency. In particular, tones at highfrequency may have a wider bandwidth than low frequency tones. Forinstance, a sound centered at 200 Hz with a width of 50 Hz may not beconsidered a tone, while a sound centered at 7000 Hz with a width of 200Hz could be considered a tone. Many sounds have no single dominantfrequency (tonal), but instead are more “noise” like. If a sound is widein bandwidth, with respect to its center frequency, then the sound isclassified as noise and may give rise to a noise masker. A noise maskerwill arise from signals that are not pure. Because noise maskers are notpure, they have a wider bandwidth and appear in many frequencies andwill mask more than the tonal masker.

[0095]FIG. 16 illustrates a tonal masker 1024 as a single vertical lineat a frequency which remains constant as the power increases to the peakpower 1026. By way of example, the tonal masker may have 46 HZbandwidth. Sounds within that bandwidth, but below the peak power level1026 are “masked.” An instrument that produces many harmonics, such as aviolin or a trumpet, may have many such tonal maskers. The method foridentifying tonal maskers and noise maskers is described in the ISODocument and the '484 patent referenced above.

[0096]FIG. 17 shows a tonal masker 1024 with its associated maskingskirt 1028. The masking skirt 1028 represents a threshold indicatingwhich signals will be masked by the tonal masker 1024. A signal thatfalls below the masking skirt 1028 (such as the signal designated 1030)cannot be heard because it is masked by the tone masker 1024. On theother hand, a smaller amplitude tone (such as tone 1032) can be heardbecause its amplitude rises above the masking skirt 1028.

[0097] As shown in FIG. 17, the closer in frequency a signal is to thetonal masker 1024, the greater its amplitude may be and still be masked.Signals that have very different frequencies from the masker 1024, suchas signal 1032, may have a lower amplitude and not fall below themasking skirt 1028, nor be masked.

[0098] Another aspect of hearing and psycho-acoustics is that a personcan hear two tones at the same frequency provided that the softer toneis close enough in amplitude to the louder tone. The maximum differencein amplitude between the two tones of common frequency is referred to asthe masking index. The masking index is dependent, in part, uponfrequency of the tones. Generally, the masking index increases withfrequency. For instance, the masking index of a masking tone at 1000 Hzwill be smaller than the masking index of a masking tone at 7000 Hz.

[0099]FIG. 17 illustrates the masking index 1034 for the tonal masker1024. The masking index 1034 is the distance from the peak 1026 of thetonal masker 1024 to the top 1036 of the masking skirt 1028. Thisdistance is measured in dB. For purposes of illustration, the graphs inFIGS. 16-19 scale the frequency along the modules of the graph in Bark.Each Bark corresponds to a frequency band distinguished by the humanauditory system (also referred to as a “critical band”). The human eardivides the discernable frequency range into 24 critical bands. Thefrequency in psycho-acoustics is often measured in Bark instead ofHertz. There is a simple function that relates Bark to Hertz. Thefrequency range of 0 to 20,000 Hertz is mapped nonlinearly onto a rangeof approximately 0 to 24 Bark, according to a known function.

[0100] At low frequencies, the human ear/brain has the ability todiscern small differences in the frequency of a signal if its frequencyis changed. As the frequency of a signal is increased, the ability ofthe human ear to discern differences between two signals with differentfrequencies diminishes. At high frequencies, a signal must change by alarge value before the human auditory system can discern the change.

[0101] As noted above, signals which lack a dominant frequency may beproduce noise maskers. A noise masker is constructed by summing all ofthe audio energy within 1 Bark (a critical band) and forming a singlediscrete “noise” masker at the center of the critical band. Since thereare 24 Bark (critical bands) then there are 24 noise maskers. The noisemaskers are treated just like the tonal maskers. This means that theyhave a masking index and a masking skirt. It is known that an audiosignal may or may not have tonal maskers 1024, but it will generallyhave 1024 noise maskers.

[0102]FIG. 18 illustrates a masking skirt 1029 similar to that describedin the ISO/MPEG Layer-II for psycho-acoustic model I. The masking skirt1029 is more complex than that of FIG. 17. The masking skirt 1029includes four mask portions 1050, 1052, 1054, and 1056, each of whichhas a different slope. The mask portions 1052-1056 are defined by thefollowing equations:

[0103] 1) Skirt Portion 1050=E;

[0104] 2) Skirt Portion 1052=F*P+G;

[0105] 3) Skirt Portion 1054=H; and

[0106] 4) Skirt Portion 1056=I−J*P,

[0107] wherein the variables E1 F, G, H, I and J representpsycho-acoustic parameters which are initially defined in preset tables,but may be adjusted by the user as explained below. The variable Prepresents the amplitude 1027 of the masker 1025 to which the maskingskirt 1029 corresponds. Thus, the slopes of the mask portions 1050-1056depend on the amplitude P of the masker 1025. The distance DZ, indicatedby the number 1053, represents the distance from the masker 1025 to thesignal being masked. As the distance DZ increased between the masker1029 and the signal to be masked, the masker 1029 is only able to coverup lower and lower amplitude signals. The masking index, AV, indicatedby the number 1055, is a function of the frequency. The masking index1055 for tonal and noise maskers are calculated based on the followingformula:

[0108] 5) AVTonal=A+B*Z; and

[0109] 6) AVNoise=C+D*Z;

[0110] wherein the variables A, B, C and D represent psycho-acousticparameters and the variable Z represents the frequency of the masker inBark. The parameters A-J and suggested values therefor have beendetermined by readily available psycho-acoustic studies. A summary ofsuch studies is contained in the book by Zweiker and Fastl entitled“Psycho-acoustics,” which is incorporated herein by reference.

[0111] ISO/MPEG LAYER-II

[0112] The CODEC 1010 utilizes the psycho-acoustical model as describedin the ISO psycho-acoustical model I as the basis for its parameters.The ISO model I has set standard values for ten model parameters (A, B,J). These model parameters are described below: A = 6.025 dB B = 0.275dB/Bark C = 2.025 dB D = 0.175 dB/Bark E = 17.0 dB/Bark F = 0.4 1/Bark G= 6.0 dB/Bark H = 17.0 dB/Bark I = 17.0 dB/Bark J − .15 1/Bark

[0113] Parameters A through J are determined as follows:

[0114] Z=freq in Bark

[0115] DZ=distance in Bark from master peak (may be + or −) as shown inFIG. 5

[0116] Pxx(Z(k))=Power in SPL(96 db=+/−32767)

[0117] at frequency Z of masker K

[0118] xx tm for tonal masker or nm for noise masker

[0119] Pxx is adjusted so that a full scale sine wave (+/−32767)generates a Pxx of 96 db.

[0120] Pxx=XFFT+96.0 where XFFT=0 db at +/−32767 amplitude

[0121] XFFT is the raw output of an FFT. It must be scaled to convert itto Pxx

[0122] AVtm(k)=A+B*Z(k) Masking index for tonal masker k

[0123] AVnm(k)=C+D*Z(k) Masking index for tonal masker k

[0124] VF(k. DZ)=E*(|DZ|−1)+(F*X(Z(k))+G)

[0125] VF(k. DZ)=(F*X(Z(k))+G)*|DZ|

[0126] VF(k. DZ)=H*DZ

[0127] VF(k. DZ)=(DZ−1)*(I−J*X(Z(k)))+H

[0128] MLxx(k, DZ)=Pxx(k)−(AVxx(K)+VF(k, DZ))

[0129] MLxx is the masking level generated by each masker k at adistance DZ from the masker.

[0130] where xx=tm or nm

[0131] Pxx=Power for tm or nm

[0132] Parameters A through J are shown in FIG. 15. Parameters A throughi are fully described in the ISO 11172-3 document.

[0133] Additional Parameters Added to ISO/XPEG LAYER-II

[0134] In addition to parameters A-J, the CODEC 1010 may use additionalparameters K-Z and KK-NN. The CODEC 1010 allows the user to adjust allof the parameters A-Z and KK-NN. The additional parameters K-Z and KK-NNare defined as follows:

[0135] Parameter K—joint stereo sub-band minimum value

[0136] This parameter ranges from 1 to 31 and represents the minimumsub-band at which the joint stereo is permitted. The ISO specificationallows joint stereo to begin at sub-band 4, 8, 12, or 16. Setting K to 5would set the minimum to 8. Setting this parameter to I would set theminimum sub-band for joint stereo to 4.

[0137] Parameter L—anti-correlation joint stereo factor

[0138] This parameter attempts to determine if there is a sub-band inwhich the left and right channels have high levels, but when summedtogether to form mono, the resulting mono mix has very low levels. Thisoccurs when the left and right signals are anticorrelated. Ifanti-correlation occurs in a subband, joint stereo which includes thatsub-band cannot be used. In this case, the joint stereo boundary must beraised to a higher sub-band. This will result in greater quantizationnoise but without the annoyance of the anti-correlation artifact. A lowvalue of L indicates that if there is a very slight amount ofanti-correlation, then move the sub-band boundary for joint stereo to ahigher valve.

[0139] Parameter M—limit sub-bands

[0140] This parameter can range from 0 to 31 in steps of 1. Itrepresents the minimum number of sub-bands which receive at least theminimum number of bits. Setting this to 8.3 would insure that sub-bands0 through 7 would receive the minimum number of bits independent of thepsycho-acoustic model. It has been found that the psycho-acoustic modelsometimes determines that no bits are required for a sub-band and usingno bits as the model specifies, results in annoying artifacts. This isbecause the next frame might require bits in the sub-band. Thisswitching effect is very noticeable and annoying. See parameter { foranother approach to solving the sub-band switching problem.

[0141] Parameter N—demand/constant bit rate

[0142] This is a binary parameter. If it is above 0.499 then the demandbit rate bit allocation mode is requested. If it is below 0.499 then thefixed rate bit allocation is requested. If the demand bit rate mode isrequested, then the demand bit rate is output and can be read by thecomputer. Also, see parameter R, operating the CODEC in the demand bitrate mode forces the bits to be allocated exactly as the model requires.The resulting bit rate may be more or less than the number of bitsavailable. When demand bit rate is in effect, then parameter M has nomeaning since all possible sub-bands are utilized and the requirednumber of bits are allocated to use all of the sub-bands.

[0143] In the constant bit rate mode, the bits are allocated in such amanner that the specified bit rate is achieved. If the model requestsless bits than are available, any extra bits are equally distributed toall sub-bands starting with the lower frequency sub-bands.

[0144] Parameter 0—safety margin

[0145] This parameter ranges from −30 to +30 dB. It represents thesafety margin added to the psycho-acoustic model results. A positivesafety margin means that more bits are used than the psycho-acousticmodel predicts, while a negative safety margin means to use less bitsthan the psycho-acoustic model predicts. If the psycho-acoustic modelwas exact, then this parameter would be set to 0.

[0146] Parameter P—joint stereo scale factor mode

[0147] This parameter ranges from 0 to 0.999999, it is only used ifjoint stereo is required by the current frame. If joint stereo is notneeded for the frame, then this parameter is not used. The parameter pis used in the following equation:

[0148] br=demand bit rate*p

[0149] If br is greater than the current bit rate (128, 192, 256, 384),then the ISO method of selecting scale factors is used. The ISO methodreduces temporal resolution and requires less bits. If br is less thanthe current bit rate, then a special method of choosing the scalefactors is invoked. This special model generally requires that more bitsare used for the scale factors but it provides a better stereo image andtemporal resolution. This is generally better at bit rates of 192 andhigher. Setting p to 0 always forces the ISO scale factor selectionwhile setting p to 0.9999999 always forces the special joint stereoscale factor selection.

[0150] Parameter Q—joint stereo boundary adjustment

[0151] This parameter ranges from −7 to 7, and represents an adjustmentto the sub-band where joint stereo starts. For example, if thepsycho-acoustic model chooses 14 for the start of the joint stereo andthe Q parameter is set to −3, the joint boundary set to 11 (14−3). Thejoint bound must be 4, 8, 12 or 16 so the joint boundary is rounded tothe closest value which is 12.

[0152] Parameter R—demand minimum factor

[0153] This value ranges from 0 to 1 and represents the minimum that thedemand bit rate is allowed to be. For example, if the demand bit ratemode of bit allocation is used and the demand bit rate is set to amaximum of 256 kbs and the R parameter is set to 0.75 then the minimumbit rate is 192 kbs (256*0.75). This parameter should not be necessaryif the model was completely accurate. When tuning with the demand bitrate, this parameter should be set to 0.25 so that the minimum bit rateis a very low value.

[0154] Parameter S—stereo used sub-bands

[0155] This parameter ranges from 0 to 31 where 0 means use the defaultmaximum (27 or 30) subbands as specified in the ISO specification whenoperating in the stereo and dual mono modes. If this parameter is set to15, then only sub-bands 0 to 14 are allocated bits and sub-bands 15 andabove have no bits allocated. Setting this parameter changes thefrequency response of the CODEC. For example, if the sampling rate is48,000 samples per second, then the sub-bands represent 750 HZ ofbandwidth. If the used sub-bands is set to 20, then the frequencyresponse of the CODEC would be from 20 to 15000 HZ (20*750).

[0156] Parameter T—joint frame count

[0157] This parameter ranges from 0 to 24 and represents the minimumnumber of MUSICAMO frames (24 millisecond for 48 k or 36 ms for 32 k)that are coded using joint stereo. Setting this parameter non-zero keepsthe model from switching quickly from joint stereo to dual mono. In theISO model, there are 4 joint stereo boundaries. These are at subband 4,8, 12 and 16 (starting at 0). If the psycho-acoustic model requires thatthe boundary for joint stereo be set at 4 for the current frame and thenext frame can be coded as a dual mono frame, then the T parameterrequires that the boundary be kept at 4 for the next T frames, then thejoint boundary is set to 8 for the next T frames and so on. Thisprevents the model from switching out of joint stereo so quickly. If thecurrent frame is coded as dual mono and the next frame requires jointstereo coding, then the next frame is immediately switched into jointstereo. The T parameter has no effect for entering joint stereo, it onlycontrols the exit from joint stereo. This parameter attempts to reduceannoying artifacts which arise from the switching in and out of thejoint stereo mode.

[0158] Parameter U—peak/rms selection

[0159] This is a binary parameter. If the value is less than 0.499, thenthe psycho-acoustic model utilizes the peak value of the samples withineach sub-band to determine the number of bits to allocate for thatsub-band. If the parameter is greater than 0.499, then the RMS value ofall the samples in the sub-band is used to determine how many bits areneeded in each sub-band. Generally, utilizing the RMS value results in alower demand bit rate and higher audio quality.

[0160] Parameter V—tonal masker addition

[0161] This parameter is a binary parameter. If it is below 0.499 the 3db additional rule is used for tonals. If it is greater than 0.499, thenthe 6 db rule for tonals is used. The addition rule specifies how to addmasking level for two adjacent tonal maskers. There is somepsycho-acoustic evidence that the masking of two adjacent tonal maskersis greater (6 db rule) than simply adding the sum of the power of eachmasking skirt (3 db). In other words, the masking is not the sum of thepowers of each of the maskers. The masking ability of two closely spacedtonal maskers is greater than the sum of the power of each of theindividual maskers at the specified frequency. See FIG. 6.

[0162] Parameter W—sub-band 3 adjustment

[0163] This parameter ranges from 0 to 15 db and represents anadjustment which is made to the psycho-acoustic model for sub-band 3. Ittells the psycho-acoustic model to allocate more bits than calculatedfor this sub-band. A value of 7 would mean that 7 db more bits (rememberthat 1 bit equals 6 db) would be allocated to each sample in sub-band 3.This is used to compensate for inaccuracies in the psycho-acoustic modelat the frequency of sub-band 3 (3*750 to 4*750 Hz for 48 k sampling).

[0164] Parameter X—adj sub-band 2 adjustment

[0165] This parameter is identical to parameter W with the exceptionthat the reference to sub-band 3 in the above-description for parameterW is changed to sub-band 2 for parameter X.

[0166] Parameter Y—adj sub-band 1 adjustment

[0167] This parameter is identical to parameter W with the exceptionthat the reference to sub-band 3 in the above-description for parameterW is changed to sub-band 1 for parameter Y.

[0168] Parameter Z—adj sub-band 0 adjustment

[0169] This parameter is identical to parameter W with the exceptionthat the reference to sub-band 3 in the above-description for parameterW is changed to sub-band 0 for parameter Z.

[0170] Parameter KK—sb hang time

[0171] The psycho-acoustic model may state that at the current time, asub-band does not need any bits. The KK parameter controls thiscondition. If the parameter is set to 10, then if the model calculatesthat no bits are needed for a certain sub-band, 10 consecutive framesmust occur with no request for bits in that sub-band before no bits areallocated to the sub-band. There are 32 counters, one for each sub-band.The KK parameter is the same for each sub-band. If a sub-band is turnedoff, and the next frame needs bits, the subband is immediately turnedon. This parameter, is used to prevent annoying switching on and off ofsub-bands. Setting this parameter nonzero results in better soundingaudio at higher bit rates but always requires more bits. Thus, at lowerbit rates, the increased usage of bits may result in other artifacts.

[0172] Parameter LL—joint stereo scale factor adjustment

[0173] If this parameter is less than 0.49999, then scale factoradjustments are made. If this parameter is 0.5000 or greater, then noscale factor adjustments are made (this is the ISO mode). This parameteris used only if joint stereo is used. The scale factor adjustmentconsiders the left and right scale factors a pair and tries to pick ascale factor pair so that the stereo image is better positioned in theleft/right scale factor plane. The result of using, scale factoradjustment is that the stereo image is significantly better in the jointstereo mode.

[0174] Parameter MM—mono used sub-bands

[0175] This parameter is identical to parameter S except it applies tomono audio frames.

[0176] Parameter NN joint stereo used sub-bands

[0177] This parameter is identical to parameter S except it applies tojoint stereo audio frames.

[0178] As the psycho-acoustic parameters affect the resultant quality ofthe audio output, it would be advantageous for users to vary the outputaccording to the user's desires.

[0179] In a preferred embodiment of the disclosed CODEC 1010, thepsycho-acoustic parameters can be adjusted by the user through a processcalled dynamic psycho-acoustic parameter adjustment (DPPA) or tuning.The software for executing DPPA is disclosed in the incorporatedSoftware Appendix and discussed in more detail below in connection withFIGS. 21-32. DPPA offers at least three important advantages to a userof the disclosed CODEC over prior art CODECS. First, DPPA providesdefinitions of the controllable parameters and their effect on theresulting coding and compression processes. Second, the user has controlover the settings of the defined DPPA parameters in real time. Third,the user can hear the result of experimental changes in the DPPAparameters. This feedback allows the user to intelligently choosebetween parameter alternatives.

[0180] Tuning the model parameters is best done when the demand bit rateis used. Demand bit rate is the bit rate calculated by thepsycho-acoustic model. The demand bit rate is in contrast to a fixed bitrate. If a transmission facility is used to transmit compressed digitalaudio signals, then it will have a constant bit rate such as 64, 128,192, 256 . . . kbs. When tuning the parameters while using the ParameterN described above, it is important that the demand bit rate is observedand monitored. The model parameters should be adjusted for the bestsound with the minimum demand bit rate. Once the parameters have beenoptimized in the demand bit rate mode, they can be confirmed by runningin the constant bit rate mode (see Parameter N).

[0181] DPPA also provides a way for the user to evaluate the effect ofparameter changes. This is most typically embodied in the ability forthe user to hear the output of the coding technique as changes are madeto the psycho-acoustic parameters. The user can adjust a parameter andthen listen to-the resulting change in the audio quality. An alternateembodiment may incorporate measurement equipment in the CODEC so thatthe user would have an objective measurement of the effect of parameteradjustment on the resulting audio. Other advantages of the disclosedinvention with the DPPA are that the user is aware of what effect theindividual parameters have on the compression decompression scheme, isable to change the values of parameters, and is able to immediatelyassess the resulting effect of the current parameter set.

[0182] One advantage of the ability to change parameters in thedisclosed CODEC, is that the changes can be accepted in real time. Inother words, the user has the ability to change parameters while theaudio is being processed by the system.

[0183] In the preferred embodiment, the compression scheme (attached asthe Software Appendix) includes thirty adjustable parameters. It iscontemplated that additional parameters can be added to the CODEC tomodify the audio output. Provisions have been made in the CODEC forthese additional parameters.

[0184] Turning now to FIG. 19, one can see two tonal maskers 1024 and1025. The individual masking skirts for these maskers are shown in 1028.The encoder predicts how do these individual maskers mask a signal inthe region in between 1024 and 1025. The summing of the masking effectsof each of the individual maskers may be varied between two methods ofsumming the effects of tonal maskers. These methods are controlled byParameter V described above.

[0185]FIG. 20 is illustrative of the steps the user must take to modifyeach parameter. As shown in FIG. 20, the parameters are set to theirdefault value (which may be obtained from one of several stored table)and remain at that value until the user adjusts the parameter. The usermay change the parameter by turning one of the knobs, pushing one key onthe keypad, or changing one of the graphics representative of one of theparameters on the computer monitor. Thus, as shown in box 1060, thedisclosed CODEC 1010 waits until the user enters a command directed toone of the parameters. The CODEC 1010 then determines which parameterhad been adjusted. For example, in box 1062 the CODEC inquires whetherthe parameter that was modified was parameter J. If parameter J was notselected, the CODEC 1010 then returns to box 1060 and awaits anothercommand from the user. If parameter J was selected, the CODEC 1010awaits for the user to enter a value for that parameter in box 1064.Once the user has entered a value for that parameter, the CODEC 1010, inbox 1066, stores that new value for parameter J. The values for thedefault parameters are stored on a storage medium in the encoder 1012,such as a ROM or other chip.

[0186] Turning again to FIGS. 14 and 15 (which generally illustrate theoperation of the disclosed CODEC)—an analog audio source 1016 is fedinto the encoder/decoder (CODEC) 1010 which works in loop back mode(where the encoder directly feeds the decoder). Parametric adjustmentscan be made via a personal computer 1040 attached to the CODEC 1010 froman RS232 port (not shown) attached to the rear of the CODEC. A cable1042 which plugs into the RS232 port, connects into a spare port (notshown) on the PC 1040 as shown in FIG. 14. The personal computer 1040 ispreferably an IBM-PC or IBM-PC clone, but can be any personal computerincluding a Macintosh™. The personal computer 1040 should be at least a386DX-33, but is preferably a 486 or better. The PC should have a VGAmonitor or the like. The preferred personal computer 1040 should have atleast 4 mb of memory, a serial com port, a mouse, and a hard drive.

[0187] Once the PC 1040 is connected to the CODEC 1010, a tuning filecan be loaded onto the personal computer 1040, and then the parameterscan be sent to the encoder via a cable 1042. A speaker 1044 ispreferably attached to the output of the CODEC 1010, via cable 1046, togive the user real time output. As a result, the user can evaluate theresults of the parameter adjustment. A headphone jack (not shown) isalso preferably included so that a user can connect headphones to theCODEC and monitor the audio output.

[0188] The parameters can be adjusted and evaluated in a variety ofdifferent ways. In the preferred embodiment, a mouse is used to move acursor to the parameter that the user wishes to adjust. The user thenholds down the left mouse button and drags the fader button to the leftor right to adjust the parameter while listening to the audio from thespeaker 1044. For example, if the user were to move the fader button forparameter J to the extreme right, the resulting audio would be degraded.With this knowledge of the system, parameter J can be moved to test thesystem to insure that the tuning program is communicating with theencoder. Once the user has changed all or some of the parameters, thenewly adjusted parameters can be saved.

[0189] in another embodiment, control knobs or a keypad (not shown), canbe located on the face of the CODEC 1010 to allow the user to adjust theparameters. The knobs would communicate with the tuning program toeffectuate the same result as with the fader buttons on the computermonitor. The attachment of the knobs can be hard with one knob allottedto each adjustable parameter, or it could be soft with a single knobshared between multiple parameters.

[0190] In another embodiment, a graphic representing an “n” dimensionalspace with the dimensions determined by the parameters could be shown onthe computer display. The operator would move a pointer in that space.This would enable several parameters to be adjusted simultaneously. Instill another embodiment, the parameters can be adjusted in groups.Often psycho-acoustic parameters only make sense when modified in groupswith certain parameters having fixed relationships with otherparameters. These groups of parameters are referred to as smart groups.Smart group adjustment would mean that logic in the CODEC would chancerelated parameters (in the same group) when the user changes a givenparameter. This would represent an acceptable surface in the adjustableparameter space.

[0191] in yet another embodiment, a digital parameter read out may beprovided. This would allow the values of the parameters to be digitallydisplayed on either the CODEC 1010 or the PC 1040. The current state ofthe CODEC 1010 can then be represented as a simple vector of numbers.This would enable the communication of parameter settings to otherusers.

[0192] Parameter adjustment can be evaluated in ways other than bylistening to the output of speaker 1044. In one embodiment, the CODEC1010 is provided with an integrated FFT analyzer and display, such asshown in applicant's invention entitled “System For Compression AndDecompression Of Audio Signals For Digital Transmission,” and theSoftware Appendix that is attached thereto, that are both herebyincorporated by reference. By attaching, the FFT to the output of theCODEC, the user is able to observe the effect of parametric changes onfrequency response. By attaching the FFT to the input of the CODEC, theuser is able to observe frequency response input. The user can thuscompare the input frequency response to the output frequency response.In another embodiment, the disclosed CODEC 1010 is provided with testsignals built into the system to illustrate the effect of differentparameter adjustments.

[0193] In another embodiment, the DPPA system may be a “teaching unit.”To determine the proper setting, of each parameter, once thedetermination is made, then the teacher could be used to disburse theparameters to remote CODECs (receivers) connected to it. Using thisembodiment, the data stream produced by the teaching unit is sent to theremote CODEC that would then use the data stream to synchronize theirown parameters with those determined to be appropriate to the teacher.This entire system thus tracks a single lead CODEC and avoids thenecessity of adjusting the parameters of all other CODECs in the networkof CODECS.

[0194] Processing Flow of the Preferred Embodiment

[0195] Next, the processing flow of the preferred embodiment isdescribed in connection with FIGS. 21-33.

[0196]FIG. 21 generally illustrates the functions of an encoder for asingle channel receiving audio signal. The encoder includes a pluralityof band pass filters separately divided into a low pass filter bank 502and a high pass filter bank 504. The low and high pass filter banks 502and 504 include a plurality of band pass filters 506. The number of bandpass filters in each filter bank may be dynamically varied during jointstereo framing by the psycho-acoustic processor as explained below. Forpurposes of illustration, four filters have been dynamically assigned tothe low pass filter bank 502, and the remaining filters have beenassigned to the high pass filter bank 504. The band pass fitters 506receive a segment of predefined length (e.g., 24 ms) of an incominganalog audio signal and pass corresponding subbands thereof. Each bandpass filter 506 is assigned to a separate pass band having-a uniquecenter frequency and a corresponding bandwidth. The widths of each passband may differ; for instance, whereby the band pass fitters for lowfrequency signals have narrower pass bands than the pass bands offilters corresponding to high frequency signals. The band pass filtersare defined such that the pass bands slightly overlap.

[0197] The subband signals output by the band pass filters 506 aredelivered to corresponding scalers 508 which adjust the gain of thesubband signals and deliver same to corresponding quantizers 510. Thesubband signals received by each scaler 508 are divided into apredetermined number of blocks (e.g. three blocks each of which is 8milliseconds in length for a 24 millisecond segment of audio data). Thescalers 508 adjust the gain of the corresponding subband signal for eachblock within a segment until the peak to peak amplitude of the subbandsignal substantially corresponds to the range of the quantizer 510. Thegain of the subband signal is controlled by the scaler 508 to ensurethat the peak to peak amplitude never exceeds the capacity of thequantizer 510. By way of example, each subband signal delivered from aband pass filter 506 may include 36 samples divided into three blocks of12 samples. The scaler 508 adjusts the gain of the 12 sample blocks asexplained above to ensure that the quantizer 510 is fully loaded. Thequantizer 510 has a maximum quantization capacity. The quantizers 510convert the incoming samples to one of a predefined number of discretelevels and outputs a corresponding digital signal representative of theclosest quantization level to the sample level. The number and distancebetween quantization levels is governed by the number of bits allocatedto the quantizer 510. For instance, the quantizer 510 will use morequantization levels if afforded 10 bits per sample as compared to thenumber of quantization levels which correspond to 6 bits per sample. Asmore bits are assigned to the quantizer, the sample is more accuratelydigitized and less noise is introduced. The quantizers 510 deliveroutput quantized subband signals to a multiplexer 512, which combinesthe subband signals to form a frame of data which is ultimatelytransmitted by the encoder.

[0198] A psycho-acoustic processor (PAP) 514 process the incoming analogaudio signal (as explained below) and controls the quantizers 510 andscalers 508 to allocate the minimum necessary number of bits to eachquantizer. In accordance with the process explained below, the PAP 514may direct the quantizer 516 to utilize six bits per sample, whilelimiting quantizer 518 to two bits per sample.

[0199]FIG. 22 generally illustrates a frame 530 having a header segment532, a data segment 534, and an ancillary data segment 536. The datasegment 534 includes multiple subband components 538 each of whichcorresponds to a unique subband (SB,-SB32). Each subband component 538is divided into three blocks 540, each of which has been scaled by thescaler 508 to properly load the quantizer 510. It is to be understoodthat the blocks 540 and subband components 538 will vary in lengthdepending upon the number of bits used by the corresponding quantizer510 to encode the corresponding subband signal. For instance, whenquantizer 516 is directed (by the path 514) to use six bits per sample,the corresponding data component 542 will include 18 bits of data (sixbits per block). However, when quantizer 518 is assigned two bits persample, data component 544 will include six bits (two bits per block).The audio data segment 534 has a fixed maximum length, and thus alimited number of bits are available for use by the quantizers 510. ThePAP 514 maximizes the bit allocation between the quantizers 510.

[0200] Once the bit allocation is complete, the PAP 514 loads thecorresponding subsection and the header segment 532 with thecorresponding encoder information 546. The encoder information 546includes the number of bits allocated to each quantizer 510 for thecorresponding subband (referred to hereafter as the “Bit AllocationInformation 548). The encoder information 546 further includes thescaling factors 550 used by the scalers 508 in connection withcorresponding blocks 540 of corresponding subband components 538. Inaddition, the encoder information 546 includes scaling factor sampleinformation 552 (explained below).

[0201]FIG. 23 illustrates an encoder including the structure of theencoder from FIG. 21, with the further ability to offer joint stereo ata decoder output. In FIG. 23, the encoder is generally denoted by block600, and the decoder is denoted by block 602. The encoder 600 receives astereo signal upon left and right channels. The decoder 602 outputs ajoint stereo signal at speakers 604 and 606. The encoder 600 includeslow pass filter banks (LPFB) 608 and 612 corresponding to the left andright channels, respectively. The encoder 600 further includes high passfilter banks (HPFB) 610 and 614, also corresponding to the left andright channels, respectively. The low and high pass filter banks 608-614include a plurality of band pass filters which are controlled by a PAP,as explained in connection with FIG. 21. The output signals of the lowpass filter banks 608 and 612 are delivered to scaler banks 616 and 618,each of which also include a plurality of scalers which operate in amanner similar to the scalers 508 in FIG. 21. The scaler banks 616 and618 deliver scaled signals to quantizer banks 620 and 622, each of whichsimilarly includes a plurality of quantizers similar to quantizers 510in FIG. 21.

[0202] While not showing, it is understood that the filter banks 616 and618 and the quantizers 620 and 622 control led by a PAP similar to thepsycho-acoustic processor 514 in FIG. 21. The low pass filter banks 608and 612, scaler banks 616 and 618, and quantizer banks 620 and 622cooperate to separately encode the lower subbands for the left and rightchannels of the stereo input signal. The encoded signals for the lowersubbands are in turn delivered from the quantizers 620 and 622 andultimately received by corresponding inverting quantizers 624 and 626.The inverting quantizers 624 and 626 cooperate with inverse scalingbanks 628 and 630 to reconvert the lower frequency portions of theencoded left and right channel signals back to analog audio.

[0203] The encoder 600 further includes a summer 632 which combines theoutput signals from the high pass filter banks 610 and 614 for the leftand right channels to produce a joint mono signal for the higher passbands. The output of the summer 632 is in turn delivered to a scalingbank 634, which scales the signal to properly load the quantizer bank636. The output signal of the quantizer bank 636 is delivered to aninverse quantizer 638 to reverse the process. The output of the inversequantizer 638 is delivered to two scaling banks 640 and 642 which arecontrolled via control channels 644 and 646.

[0204] The encoder 600 further includes calculating modules 650 and 652,which measure the energy in the corresponding high pass subbands. Themodules 650 and 652 then adjust the gain of scalers 640 and 642 inproportion to the energy of the corresponding high pass subbands. Forinstance, if HPFB 610 outputs more energy than HPFB 614, then scaler 640is set to boost the gain of its input signal greater than the gain boostof scaler 642. Thus, the audio signal in the higher pass bands is outputpredominantly at - speaker 604. The energy calculator 650 and 652 may becarried out by the psycho-acoustic processor in a manner explainedbelow.

[0205] Next, the discussion turns to the process followed by the presentinvention to undergo encoding.

[0206] With reference to FIG. 24, the PAP 514 cooperates with thequantizer 510 and scaler 508 to digitize the analog audio signalsreceived from each band pass filter 506 for corresponding subbands (step2400). In step 2402, the digitized signals for the subbands from eachbandpass filter are divided into a predefined number of blocks. Forexample, a 24 millisecond segment of analog audio may be converted to 36digital samples and then divided into three blocks of 12 samples each.In step 2404, each block of samples is analyzed to determine the maximumamplitude of the digitized signal therein. In step 2406, the scalers 508are adjusted to vary the scale of the samples within each block untilthe samples correspond to a signal gain substantially equaling the rangeof the quantizers 510.

[0207] Turning to FIGS. 25A and 25B, while the scalers 508 are beingadjusted (as explained in connection with FIG. 24), the PAP 514calculates the global masking threshold (GMT) to be used in connectionwith the present sample of analog audio data. Beginning at step 2502,the PAP 514 obtains a working table of psycho-acoustic parameters havinga value for each of parameters A-NN (described above). The table ofparameters may be one of several predefined tables stored in memory inthe encoder . . . The table is updated dynamically by the user duringoperation of the encoder. For instance, when the encoder is initiallystarted, an initial set of parameter values may be read from the encodermemory and used to initialize the encoder. Thereafter, as the PAP 514continuously processes segments of analog audio data, the user may varythe parameter values stored in the working table. Once the user varies aparameter value in the working table, the PAP 514 obtains the newparameter value set for processing the following analog audio segments.For instance, after the user listens to a short segment (one minute) ofanalog audio encoded and decoded according to the initial working table,the user may desire to adjust the parameters within the working table.Once the user adjusts these parameters, the PAP 514 effects subsequentpsycho-acoustic processing based on the new parameter values assigned bythe user. Thus, the user is afforded the opportunity to listen to thesignal which results from the users adjustment in the parameters.

[0208] Returning to FIG. 25A, once the PAP 514 obtains the working tableof parameters A-NN, the PAP 514 uses these parameter values for thecurrent segment of audio data. At step 2504, the PAP 514 obtains asegment of analog audio data of ;predetermined length (e.g., 24milliseconds). The segment is digitized. At step 2506, the PAP 514converts the digitized segment from the time to the frequency domainaccording to the bark scale. These conversions may be effected using aFast Fourier Transform and a known Bark transfer function between thebark frequency domain and the normal frequency domain. At step 2508, thePAP calculates the threshold of hearing. At step 2510, the PAP analyzesthe signal converted in step 2506 to the bark frequency domain to locatethe tonal peaks therein. Once located, the tonal peaks are removed instep 2512 from the digital converted signal. Next, the digitized signalis divided into critical bands (step 2514). Noise maskers are calculatedfor each critical band by summing the remaining energy within eachcritical band (after the tonal peaks have been removed). Arepresentative noise masker is obtained for each critical band from thenoise calculated in step 2514. It is understood that, a signal noisemasker is substituted therefore at a single frequency and having apredetermined amplitude. The amplitude and frequency of the noise maskerare determined by the amount of noise energy within the critical band.

[0209] At step 2516 (FIG. 25B), the PAP calculates masking skirts forthe tonal and noise maskers based on parameters A-J and based on theamplitudes and frequencies of the tonal and noise maskers. At step 2518,the PAP combines the noise and tonal masking skirts and the threshold ofhearing to obtain a global masking threshold for the presently digitizedsegment of audio=data. The global masking threshold (GMT) is dividedinto subbands at step 2520. The subbands correspond to the band passfilters 506. At step 2520 the PAP locates the maximum and minimum ofeach global masking threshold within each subband. At step 2522 the PAPassigns quantization levels for each subband based on amount of noisewhich may be added to each subband without exceeding the minimum valueof the GMT within the corresponding subband. The assignment process isdescribed in more detail below.

[0210] Turning to FIG. 26, the process of obtaining the GmT is explainedin more detail. At step 2600, the PAP locates the first subband (subband0) and obtains the first masker within this subband (step 2602). At step2604, the PAP combines the current masker obtained in step 2602 with thethreshold of hearing to obtain an initial GMT for the subband.Thereafter the next masker is obtained at step 2606. The PAP thendetermines at step 2608 whether the newly obtained and preceding maskersrepresent adjacent tonal maskers. If two adjacent tonal maskers arebeing combined, control flows to step 2610 at which the PAP combines thetwo adjacent total maskers within the GMT using one of two additionrules defined by parameter V. For instance, the two tonal maskers may becombined according to a 3 db or a 6 db addition rule based upon which ischosen by the parameter V. The tonal maskers are combined according toone of the following equations:

[0211] 3 db(rule)=10 log12(10 P_(1 (db))/10+10P2_(2 (db))/10)

[0212] 6 db(rule)=2 log12(1 P_(1 (db))/2+1P_(2(db))/2)

[0213] Returning to step 2608 if the two maskers are not tonal maskers,flow moves to step 2612 at which the maskers are combined with theglobal masking threshold according to the conventional method. Next, atstep 2614 it is determined whether the current masker represents thelast masker in the subband. If not, steps 2606-2612 are repeated. If thecurrent masker represents the last masker in the subband, flow passes tostep 2616 at which the PAP determines whether the current subband is oneof subbands 0, 1, 2 and 3, if so, control passes to step 2618 at whichthe global masking threshold for the current subband is adjusted by abiasing level determined by the corresponding one of parameter W-Z. Forinstance, if the current subband is subband 2, then the GMT withinsubband 2 is adjusted by a db level determined by parameter Y. At step2620 it is determined whether the last subband has been analyzed. Ifnot, flow pass to step 2602 where the above described processesrepeated. Otherwise, control returns to the main routine illustrated inFIG. 25.

[0214] Next, the quantization level assignment process of step 2522 isdescribed in more detail in connection with FIG. 30. The assignmentprocess involves three primary phases, namely an allocation phase, adeallocation phase and an excess bit allocation phase. During theallocation phase step (3000), the PAP steps through each subband foreach channel (left and right) and assigns the corresponding quantizer anumber of bits to be used for quantizing the subband signal. During bitallocation, the number of bits allocated to a subband are incremented inpredefined allocation steps until a sufficient number of bits areassigned to the quantizer to ensure that the noise introduced into thesignal during the quantizing process is below the minimum of the GMT forthe subband. Once the necessary number of bits are assigned to eachsubband at step 3000 it is determined whether the number of bitsallocated has exceeded the number of bits available (i.e., the bit pool)at step 3002. If not, and extra bits exist then control flows to step3004. At step 3004, the PAP determines whether the encoder is operatingin a demand or constant bit rate mode. In a demand mode, once the PAPallocates bits to each subband, the allocations become final, eventhrough the total number of bits needed is less than the numberavailable for the current transmission rate. Thus, the allocationroutine ends. However, when in a constant bit rate mode, the extra bitsare distributed evenly or unevenly among the subbands.

[0215] It is desirable to choose the demand bit rate made when tuningthe codec to ensure that the signal heard by the user accuratelyreflects the parameter values set by the user. The remaining bits fromthe bit pool are distributed amongst the subbands to further reduce thequantization noise. However, if bit allocation in step 3000 has exceededthe bit pool then flow passes to step 3006 at which bit deallocation isperformed and previously assigned bits are removed from selectedquantizers which are deemed the best candidate for deallocation.Deallocation occurs with respect to those subbands at which deallocationwill have the least negative effect. Put another way, the PAPdeallocates bits from subbands which will continue, even afterdeallocation, to have quantization noise levels closest to the GMTminimum for that subband (even though the quantization noise levelexceeds the GMT minimum).

[0216] During bit allocation, flow passes at step 3000 to the routineillustrated in FIG. 27. At step 2702, the PAP determines whether theencoder is operating in a stereo, mono, or joint stereo framing mode.The PAP sets the last subband to be used which is determined by thesubband limit parameters S, MN and NN. At step 2704, the PAP determinesthe total number of bits available (i.e., the bit pool) for the currentframing mode, namely for joint stereo, stereo or mono. At step 2706, thefirst subband and first channel are obtained. At step 2708, the maximumfor the signal within the current subband is compared to the GMT minimumwithin the current subband. If the subband signal maximum is less thanthe GMT minimum, then the current subband signal need riot necessarilybe transmitted since it falls below the GMT. Thus, flow passes to step2710 at which it is determined whether the current subband falls below asubband limit (defined by parameter M). If the current subband is belowthe subband limit then the PAP allocates bits to the subband eventhrough the subband signal falls below the GMT minimum. For instance, ifthe current subband is two and the user has designated (via parameter M)that subbands 0-5 should be encoded and transmitted, then subband 2would be encoded by the corresponding quantizer with a minimum number ofbits allocated to the quantizer. Thus, at step 2710, if the currentsubband is less than the subband limit then control passes to step 2712at which the bit allocation routine is called to assign at least a firstallocation step of a minimum number of bits to the current subband.However, at step 2710 if it is determined that the current subband isgreater than the subband limit then control passes to step 2718 and thebit allocation routine is bypassed (i.e. the quantizer for the currentsubband is not assigned any bits and thus the signal within the currentsubband is not encoded, nor transmitted). At step 2712, prior toperforming the bit allocation routine, the digitized audio signal%within the current subband is adjusted to introduce a safety margin orbias thereto to shift the digitized signal upward or downward. Thissafety margin represents a parameter adjusted dynamically by the user(parameter 0).

[0217] After flow returns from the bit allocation routine, it isdetermined at step 2714 whether the encoder is operating in a jointstereo mode. If not flow passes to step 2718 at which it is determinedwhether the foregoing process (steps 2708-2714) need to be repeated forthe opposite channel. If so, the channels are switched at step 2724 andthe process is repeated. If not, flow passes from step 2718 to 2722 atwhich it is determined whether the current subband is the last subband.If not, the current subband is incremented at step 2726 and theallocation routine is repeated. Thus, steps 2708-2726 are repeated eachsubband.

[0218] Returning to step 2714, when operating in a joint stereo mode,control passes to step 2716 at which it is determined whether the bitallocation routine at step 2712 allocated a number of bits to thecurrent subband which resulted in the total number of allocated bitsexceeding the available bit pool for the current mode. If so, thecurrent subband number is recorded at step 2720 as the subband at whichthe bit pool boundary was exceeded.

[0219] When in a stereo mode the process flows from step 2708 to step2726 without using steps 2716 and 2720 in order that every subbandwithin the right and left channels is assigned the necessary number ofbits to insure that the quantization noise falls below global maskingthreshold within the corresponding subband. When in the joint stereomode, the foregoing process is repeated separately for every subbandwithin the left and right channels (just as in the stereo mode).However, the system records the subband number at which the availablebit pool was exceeded in step 2720. This subband number is later used todetermine a joint stereo boundary such that all subbands below theboundary are processed separately in stereo for the left and rightchannels. All subbands above the boundary are processed jointly, such asshown by the joint stereo encoder of FIG. 23. The subband boundarycorresponds to the break point between the low pass filter banks 608 and612 and the high pass filter banks 610 and 614 (shown in FIG. 23).

[0220] Turning to FIG. 28, the bit allocation routine is described inmore detail. Beginning at step 2802, an array of allocation steps isobtained for the current mode (e.g., stereo, mono or joint stereo. Eachlevel within the array corresponds to a predefined number of bits to beassigned to a quantizer. By way of example, the array may include 17elements, with elements 1, 2 and 3 equaling, 60 bits, 84 bits and 124bits, respectively. Thus, at the first step 60 bits are assigned to thequantizer corresponding to the current subband. At the second step, 84bits are assigned to the quantizer corresponding to the current subband.Similarly, at the third step, 124 bits are assigned to the quantizer forthe current subband. The steps are incremented until the current stepallocates a sufficient number of bits to the quantizer to reduce thequantization noise below the minimum GMT for the current subband. Inaddition to the bit allocation array, a mask to noise ratio array isincluded containing a list of elements, each of which corresponds to aunique step. Each element contains a predefined mask to noise ratioidentifying the amount of noise introduced into the encoded signal whena given number of bits are utilized to quantize the subband. Forinstance, steps 1, 2 and 3 may correspond to mask to noise ratios (MNR)of 10 db, 8 db and 6 db, respectively. Thus, if 60-bits are allocated tothe current quantizer for quantizing the current subband, lodb of noisewill be introduced into the resultant encoded signals. Similarly, if 84bits are used to quantize the signal within the current subband, 8 db ofnoise are introduced.

[0221] At step 2802, the allocation and MNR arrays are obtained and thecurrent step is set to 1. At step 2804, the allocation array is accessedto obtain the number of bits to be allocated to the current subband forthe current step. At step 2806 the maximum level of the audio signalwithin the current subband is obtained based on one of the audio peak orRMS value, which one selected between determined by parameter U. Next,the MNR value for the current step is obtained from the MNR array(2808). At step 2810, it is determined whether the audio signal maximum,when combined with the MNR value of the current allocation step, exceedthe minimum of the GMT for the current subband. If so, then a detectableamount of noise will be introduced into the signal if the currentallocation step is used. Thus, control passes to step 2816.

[0222] At step 2816, the PAP records the difference between the GMTminimum of the current subband and the level combined signal formed fromthe maximum value for the audio signal and the MNR. Thereafter, at 2818the allocation step is incremented in order to allocation more bits tothe current subband. The foregoing loop is repeated until the allocationstep is incremented sufficiently to allocate a number of bits to thecurrent subband necessary to reduce the combined signal formed from theaudio signal max and MNR below the minimum of the GMT. Once it isdetermined at step 2810 that this combined signal is less than theminimum of the GMT, control passes to step 2812. At step 2812, thenumber of bits corresponding to the current step are allocated to thequantizer for the current subband. At step 2814, the system updates thetotal number of allocated bits for the current segment of audioinformation.

[0223] According to foregoing process, each quantizer is assigned anumber of bits corresponding to an allocation step which is justsufficient to reduce the combined noise and audio signal below theminimum of the GMT. In addition, at step 2816, the system retains adeallocation table having one element for each subband and channel. Eachelement within the table corresponds to the difference between the GMTminimum and the combined audio signal maximum and MNR value for theallocation step preceding the allocation step ultimately assigned to thequantizer in step 2812.

[0224] B way of example, a quantizer may be assigned the number of bitscorresponding to allocation step 3 (e.g., 124 bits). At step 2816, itwas determined that the signal and MNR for step 2 exceeded the GMTminimum by 3 db. The deallocation table will record at step 2816 this 3db value indicating that, while the current quantizer is assigned toallocation step 3, if the current quantizer had been assigned toallocation step #2, the combined signal and MNR would exceed the GMTminimum by 3 db. The deallocation table recorded at step 2816 may beused later if the deallocation of bits becomes necessary (as explainedbelow).

[0225] The bit allocation routine of FIG. 28 is continuously repeatedfor each channel and for each subband (according to the process of FIG.27). Once control returns to step 3000 in FIG. 30, all of the subbandsfor- both channels have been allocated the necessary number of bits. Atstep 3002 if it is determined that the number of bits allocated exceedsthe bit pool, control passes to step 3006 which is illustrated in moredetail in FIG. 31.

[0226] When it is determined that deallocation is necessary, controlpasses from step 3006 (FIG. 30) to the deallocation routine illustratedin FIG. 31. At step 3102, it is determined whether the encoder isoperating in a joint stereo mode. If so, control passes to step 3104 atwhich the joint stereo boundary is determined. The joint stereo boundaryrepresents the boundary between the low pass filter banks 608 and 612and high pass filter banks 610 and 614 (FIG. 23). Subbands below thejoint stereo boundary are processed separately for the left and rightchannels within the low pass filter banks 608 and 612. Subbands abovethe joint stereo boundary are included within the high pass filter banks610 and 614 and are combined in summer 632 to form a mono signal. Thus,subbands above the joint stereo boundary are combined for the left andright channels and passed through a single quantizer bank 636.

[0227] Returning to FIG. 31, once the joint stereo boundary isdetermined, a new bit pool is obtained based on the joint stereoboundary (step 3106). A new bit pool must be calculated since theoriginal bit pool which calculated based on full stereo whereby it waspresumed that bits would be allocated to all of the subbands separatelyfor the left and right channels. However, subbands above the boundaryare combined for the left and right channels and thus additional bitsare available for allocation. For instance, in a full stereo systemusing 22 subbands per channel, bits must be allocated between 44separate subbands (i.e., 22 subbands for the left channel and 22subbands for the right channel). However, in a joint stereo modeutilizing, 22 subbands with the joint stereo boundary at subband 8, only32 subbands are necessary (i.e., eight lower subbands for the leftchannel, eight lower subbands for the right channel and 16 uppersubbands for the combined signals from the left and right signals). Oncethe new bit pool is calculated, the joint stereo array is obtained atstep 3108. The joint stereo array identifies the allocation stepscombining the number of bits to be allocated for each step during thebit allocation routine (FIG. 28). In addition, the joint stereo arrayidentifies the mask to noise ratio for each allocation step. At step3110, the bit allocation routine (FIG. 28) is called to allocate bits tothe subbands, wherein subbands below the joint stereo boundary areseparately allocated for the left and right channels, while subbandsabove the joint stereo boundary are allocated for a single set of bandpass filters representing the combination of the signals from the leftand right channels.

[0228] Next, at step 3112, it is determined whether the bit allocationfor the joint stereo frame exceeds the joint stereo bit pool (obtainedat step 3106). If not, control returns to the routine in FIG. 30.However, if -more bits have been allocated than are available in the bitpool, control passes to step 3114 to begin a deallocation process. Atstep 3114, the deallocation table (generated at step 2816 in FIG. 28) issorted based on the difference values recorded therein to align thesedifference values in descending order. At step 3116, the first elementwithin the deallocation table is obtained. At step 3118, a deallocationoperation is effected. To deallocate bits, the quantizer correspondingto the channel and subband identified in the first element of thedeallocation table is assigned a new number of quantizing bits. Thenumber of bits newly assigned to this quantizer corresponds to the steppreceding the step original assigned to the quantizer. For instance, ifduring the original allocation routine, a quantizer was assigned 124bits (corresponding to step 3), then at step 3118, the quantizer wouldbe assigned 84 bits (corresponding to allocation step 2).

[0229] At step 3120, a new difference value is calculated for thecurrent subband based on the allocation step preceding the newlyassigned allocation step. This new difference is added to the differencetable at step 3122. The number of deallocated are then subtracted fromthe allocated bit total (step 3124). Thereafter, it is determinedwhether the new total of bits allocated still exceeds the available bitpool (step 3126). If not, control returns to step 3006 (FIG. 30). If theallocation bit total still exceeds the bit pool, control returns to step3114 and the above described deallocation processes is repeated.

[0230]FIGS. 32 and 33 set forth an example explained hereafter inconnection with the allocation steps and deallocation routine. FIGS. 32Aand 32B illustrate two exemplary subbands with the correspondingportions of the global masking threshold and the quantized signal levelsderived from the audio signal peak and MNR value. The quantized signallevels are denoted at points 31063108 and 3110-3113. The minimums of theGMT are denoted at levels 3204 and 3205. Stated another way, if thenumber of bits associated with allocation step #1 are assigned to thequantizer for subband 3 (FIG. 32A-), the resultant combined audio signaland MNR will have a magnitude proximate line 3206. If more bits areassigned to the quantizer (i.e., allocation step #2), the combinedsignal and MNR value is reduced to the level denoted at line 3207.Similarly, at allocation step #3, if additional bits are allocated tothe quantizer the combined audio signal and MNR value will lie proximateline 3208.

[0231] With reference to FIG. 32B, at allocation step #1 the combinedaudio and MNR level will lie proximate line 3210. At step #2, the itwill be reduced to level 3211, and at allocation step #3, it will fallto line 3212. At allocation step 4, sufficient bits will be allocated tothe quantizer to reduce the combined signal and MNR value to level 3213which falls below the GMT min at point 3205.

[0232] The bit allocation routine as discussed above, progresses throughthe allocation steps until the combined signal and MNR value (hereafterthe quantizing valve) falls below the minimum of the GMT. During eachinnervation through the bit allocation routine, when the quantizingvalue is greater than the GMT min, the deallocation table is updated toinclude the difference value between the minimum of the GMT and the MNRvalue. Thus, the deallocation table of FIG. 32 stores the channel andsubband for each difference value. In the present example, thedeallocation table records for subband 3 (FIG. 32A) the difference value3 db which represents the distance between the minimum of the GMT atpoint 3204 and the quantization level at point 3207 above the GMT. Thetable also stores the allocation step associated with the quantizationvalue at line 3207. The deallocation table also stores an element forsubband 7 which represents the difference value between the minimum ofthe GMT and the quantization level corresponding to line 3212.

[0233] During the deallocation routine, the deallocation table isresorted to place with the difference values in ascending order, suchthat the first element in the table corresponds to the subband with theleast difference value between the minimum GMT and quantization level ofthe next closest MNR value. The quantizer corresponding to subband 7 isdeallocated, such that the number of bits assign thereto is reduced fromthe number of bits corresponding to step #4 (line 3213) to the number ofbits corresponding to step #3 (line 3212). Thus, the deallocationroutine subtracts bits from the subband which will introduce the leastamount of noise above the GMT for that subband. Once the subband 7 hasbeen deallocated, the difference value is recalculated for the nextpreceding step (corresponding to MNR at line 3211). This new differencevalue is stored in the deallocation table along with its correspondingallocation step. If the number of bits deallocated during the first passthrough this process is insufficient to lower the total allocated bitsbelow the available bit pool maximum, than the processes repeated. In asecond innervation, the quantizer corresponding to subband 3 would bereallocated with fewer bits corresponding to allocation step #2 (line3207). This process is repeated until the total allocated bits fallswithin the available bit pool.

[0234] Basic components and CODEC System

[0235]FIG. 1 illustrates a high level block diagram of a CODEC 1. FIG. 1shows an encoder digital signal processor (DSP) 1, a decoder DSP 2, anLED DSP 95, an asynchronous multiplexer 3, an asynchronous demultiplexer6, at least one digital interface module (DIM) 7 connected to theencoder output, at least one DIM 8 connected to the decoder input, aloopback control module 9, and a control processor 5. The encoder 1inputs- digital signals and timing signals and outputs compressed audiobit streams. The decoder 2 similarly inputs compressed audio bit streamsand timing signals and outputs decompressed digital signals.

[0236] The CODEC 1 is capable of holding several audio compressionalgorithms (e.g. ISO MPEG and G0.722). These and other algorithms mightbe downloaded into the CODEC from ISDN and thus future upgrades aresimple and effortless to install. This creates an extremely versatileCODEC that is resistant to obsolescence. This should be contrasted tothe ROM type of upgrade procedure currently employed by most CODECmanufacturers.

[0237] The CODEC 1 may also use a unique compression technique which isexplained below and is described in the attached Software Appendix. Thiscompression technique also uses an increased number of psycho-acousticparameters to facilitate even more efficient compression anddecompression of audio bit streams. These additional parameters aredescribed above.

[0238] The CODEC 1 also contains a control processor 5 for receiving andprocessing control commands. These commands are conveyed to the variousCODEC 1 components by a line 51. These commands might be entered by auser via front panel key pads such as 15, 152, and 154, as shown inFIGS. 5, 6, and 7. Keypad commands enter processor 5 through a line 52.The keypad also allows the user to navigate through a menu tree ofcommand choices which fall into the general categories of commoncommands, encoder commands, decoder commands, and maintenance commands.Such menu choices are displayed on a Front Panel LCD display (not shown)via signals from a processor 5 on a line 58. (See LCD Menu Summary ofcommands, Chap 8 of CODEC manual, attached to the end of thisspecification before the claims). The LCD display might also be used forcharacters to show responses to front panel user commands as well asspontaneous messages such as incoming call connect directives.Additionally, the LCD display may be used to display graphicalinformation.

[0239] The CODEC processor 5 may receive commands from a front panelremote control panel (RS232 interface format) and enter the processor 5through the line 54. A front panel remote control allows computer accessto all internal functions of the CODEC 1. Front panel remote control isespecially useful for applications that need quick access via a palm topor lap top computer. This frequently occurs in control rooms where thereare many CODECs in equipment racks serving different functions. A fullcomplement of remote control commands exists to facilitate control ofthe CODEC 1 (See the listing of remote control commands from the“cdqPRIMA” operating manual. Chapter 9, attached to the end ofspecification).

[0240] Referring again to FIG. 2, this more detailed block diagram ofCODEC 1 shows external front panel remote control data interacting withFront Panel Remote Control UART 178 via a line 54. UART 178 iscontrolled by the Control Micro 5 via a control network line 155.

[0241] The CODEC 1 also provides a rear panel remote control port whichuses either RS232 or RS485 interface formats. The RS485 port may beeither a 2 or 4 wire interface. A rear panel remote control also allowscomputer access to all the internal functions of the CODEC 1. Rear panelremote control is especially useful for applications which needpermanent access to the CODEC 1 via computer control. This frequentlyoccurs when the CODEC 1 is remotely located from the control room. Theelectrical interface choice is controlled by a command entered throughremote control or a keypad.

[0242] Referring again to FIG. 2, this more detailed block diagram ofthe CODEC 1 shows external rear panel remote control data interactingwith Remote Control UART 18 via line 56. UART 18 is controlled byControl Micro 5 via the control network line 155.

[0243] The CODEC also includes a Front Panel LED display 3, examples ofwhich are shown in FIGS. 11 and 12. This includes a set of Status,Encoder, and Decoder LED's to show the status of various CODECfunctions, for instance which compression algorithm is being used,and/or whether error conditions exist. The Status 31, Encoder 32, andDecoder 33 groups of LED's might be independently dimmed to allowemphasis of a particular group.

[0244] Referring again to FIG. 1, signals from control processor 5 enterLED DSP 95 through the line 51. These control signals are processed by aLED DSP 95 and drive a LED display 3 (FIGS. 11 and 12) via a line 96.

[0245] A LED display 3 also shows peak and average level indications forthe encoder 32 (left and right channels) and the decoder 34 (left andright channels). Each LED represents −2 dB of signal level and themaximum level is labeled dB. This maximum level is the highest levelpermissible at the input or at the output of the CODEC. All levels aremeasured relative to this maximum level. The level LED's display a 4 dBaudio range. A peak hold feature of the level LED's shows the highestlevel of any audio sample. This value is instantly registered and thesingle peak level LED moves to the value representing this signal. Ifthe peak level of all future signals are smaller, then the peak LEDslowly decays to the new peak level. The peak level LED utilizes a fastattack and slow decay operation. The LED display 3 also includes a leveldisplay to show stereo image 36 which is used to display the position ofthe stereo image. This is useful when setting the levels of the left andright channels to insure the proper balance. Also included is acorrelation level display 38 which is used to check if the left andright channels are correlated. If the left and right channels arecorrelated, then they might be mixed to mono. The level LED's might alsobe used to display a scrolling message.

[0246] Referring again to FIG. 2, this more detailed block diagram ofCODEC 1 shows the LED DSP 95 driving a LED Array 125 via a connection96. As also shown, the LED DSP 95 is controlled by the Control Micro 5via the control network line 155. The DSP 95 also drives an Headphone(Hp) D/A Converter 98 via a connection 97. A converter 98 then outputsthis analog signal via a connector 99 to external headphones (notshown). The headphones allow the user to monitor both the input andoutput signals of the CODEC 1. FIGS. 11 and 12 show headphone indicators31 at the far right of the level displays to denote the signal output tothe headphones. If both LED's are illuminated, then the left audiochannel is output to the left earphone and the right audio channel isoutput to the right earphone. If only the left LED is illuminated, theleft audio channel is output to both the left and right headphone.Similarly, if only the right LED is illuminated, the right audio channelis output to both the left and right headphone.

[0247] Analog inputs and Outputs

[0248]FIG. 2 shows a more detailed block diagram of the CODEC 1structure. Ref erring to FIGS. 1 and 9, the left audio signal 12 and theright audio signal 14 are external analog inputs which are fed into anAnalog to Digital (A/D) Converter 1, and converted into digital signalson a line 11. Similarly digital audio output signals on a line 121 areconverted from Digital to Analog (D/A) via a converter 15. Theconverters 1 and 15 use an 18 bit format. The analog sections of theCODEC are set to +18 dBu maximum input levels, other analog input andoutput levels might used.

[0249] Direct Digital Inputs and Outputs

[0250] Referring again to FIG. 1, the CODEC 1 also allows for directinput of digital audio information via an AES/EBU digital audiointerface on line 16 into encoder 1. The decoder 2 similarly outputsdecoded, decompressed digital audio information on AES/EBU output line22. These interfaces allow for interconnection of equipment without theneed for A/D conversions. It is always desirable to reduce the number ofA/D conversions since each time this conversion is performed, noise isgenerated. These interfaces might use a DB9 or XLR connectors.

[0251] AES/EBU digital input and output rates might vary and thereforesuch rates are converted, or adapted, by a Sample Rate Converter 11, toeliminate any digital clock problems. The A/D Converter 1 signals aresimilarly converted, or adapted, by a Sample Rate Convertor 11 beforeentering the encoder 1. Because of the rate adapters, the input/outputdigital rates are not required to be the same as the internal rates. Forexample, it is possible to input 44.1 kHz AES/EBU digital audio inputand ask the CODEC 1 to perform compression at 48, 44.1 or 32 kHz (byusing the front panel LCD display or a remote control command). This ispossible because of the digital rate adapters, similarly, digital audioinput sources might be 32, 44.1, or 48 kHz. These input sampling ratesare automatically sensed and rate adapted. The compression technique atthe encoder determines the internal digital sampling rate at thedecoder, and a control command is used to set this rate. The AES/EBUdigital output sampling rate from the decoder is also set via a controlcommand and might be a variety of values.

[0252] The digital audio is output from the decoder at the sampling ratespecified in the header. This rate might then be converted to otherrates via the Sample Rate Convertor 12. The Sample Rate Convertors 11,12 are capable of sampling rate changes between 0.51 and 1.99. Forexample, if the receiver received a bit stream that indicated that thesampling rate was 24 kHz, then the output sampling rate could be set to32 or 44 kHz but not 48 kHz since 48 kHz would be a sampling rateconversion of 2 to 1. This is out of the range of the sampling rateconverter. The allowed output sampling rates include 29.5, 32, 44.1, and48 kHz. Other direct digital I/O formats might include, for example,SPDIF or Optical.

[0253] The encoder 1 receives direct digital input via a connector onthe rear panel (line 16) Analog or digital signals (but not bothsimultaneously) may be input into the CODEC 1 as selected by a frontpanel switch. If the digital input is selected, the CODEC 1 locks to theincoming AES/EBU input and displays the lock condition via a front panelLED. If digital audio input is selected, an AES phase-lock loop (PLL) isused to lock onto the signal. Accordingly, the AES PLL lock light mustbe illuminated before audio is accepted for encoding. In normaloperation, the CODEC 1 locks its internal clocks to the clock of thetelephone network. For loopback (discussed below), the CODEC 1 locks itsclocks to an internal clock. In either case, the clock used by the CODEC1 is not precisely the same frequency as the AES/EBU input. To preventslips from occurring due to the presence of two master clocks, a ratesynchronizer is built into the encoder section to perform the necessaryrate conversion between the two clocks.

[0254] The decoder 2 outputs direct digital signals via a rear panelconnector (line 22). Additionally, the decoder may be synchronized to anexternal clock by an additional connector (SYNC, line 24) on the rearpanel. Referring also to FIG. 8, a block diagram is shown of the decoderoutput timing with the AES/EBU SYNC (line 24) disabled or not presentduring normal timing. If no input is present on the decoder AES/EBU SYNCinput line 24 (FIG. 1), then the output AES/EBU digital audio isgenerated by the internal clock source 2 that is either at the telephoneor internal clock rate. FIG. 9 additionally shows a block diagram of thedecoder output timing with the AES/EBU SYNC disabled or not present, andusing internal crystal timing.

[0255] Referring to FIG. 1, a block diagram is shown of the decoderoutput timing with the AES/EBU SYNC (line 24) enabled and present usingAES-timing. If the SYNC input is present, then the digital audio outputis generated at the frequency of the SYNC input via the clock generator25 being fed into the rate adaptor 252. This adapted rate is used by theD/A Converter 254, as well as the AES/EBU transmitter and receiver units256., 258. The presence of a valid sync source is indicated byillumination of the front panel AES PLL LED. The sync frequency many beslightly different from that of the CODEC 1 clock source and again therate synchronism is performed to prevent any undesired slips in thedigital audio output. The SYNC input is assumed to be an AES/EBU signalwith or without data present. The CODEC 1 only uses framing forfrequency and sync determination.

[0256] Referring again to FIG. 2, this more detailed block diagram ofCODEC 1 shows external digital input 16 entering AES/EBU receiver 13.The receiver output 14 then enters the Sample Rate Converter 11 and therate is converted, if necessary, as described above. The converter 11then feeds the rate adjusted bit stream via a line ill into the encoder1 for coding and compression.

[0257] Conversely, FIG. 2 also shows the Decoder DSP 2 outputting adecoded and decompressed bit stream via a line 123 into the Sample RateConverter 12. The converter 12 adapts the rate, if necessary, asdescribed above and outputs the rate adjusted bit stream via line 122into a AES/EBU Transmitter 126. The transmitter 126 then outputs thedigital signal through an external connection 22.

[0258]FIG. 2 also shows the AES/EBU digital synchronous input line 24leading into a AES/EBU Receiver 146.. The receiver 146 routes thereceived SYNC input data into the Sample Rate Converter 12 via a line147. The converter 12 uses this SYNC input for rate adapting asdescribed above.

[0259] Asynchronous Ancillary Data

[0260] The CODEC 1 is also capable of handling a variety of ancillarydata in addition to primary audio data. The audio packet, for instance,consists of a header, audio data, and ancillary data. If the samplingrate is 48 KHz, then the length of each packet is 24 milliseconds. Theheader consists of a 12 bit framing pattern, followed by various bitswhich indicate, among other things, the data rate, sampling rate, andemphasis. These header bits are protected by an optional 16 bit CRC. Theheader is followed by audio data which describes the compressed audiosignal. Any remaining bits in the packet are considered ancillary data.

[0261] Referring again to FIG. 1, the CODEC 1 provides for transmissionof ancillary data via an asynchronous, bidirectional RS-232 inputinterface 39, and an output interface 62. These interfaces provide atransparent channel for the transmission of 8 data bits. The data formatis 1 start bit, 8 data bits, 1 stop bit and no parity bits. A maximumdata rate might be selected by the control processor 5. This interfaceis capable of transmitting at the maximum data rate selected for theencoder 1 and the decoder 2 and thus no data pacing such as XON/XOFF orCTS/RTS are provided.

[0262] The RS-232 data rates might be set from 3 to 19,2 bps. The use ofthe ancillary data channel decreases the number of bits available to theaudio channel. The reduction of the audio bits only occurs if ancillarydata is actually present. The data rate might be thought of as a maX3data rate and 3 f there is no ancillary data present, then no ancillarydata bits are transmitted. A typical example of this situation occurswhen the CODEC 1 is connected to a terminal; when the user types acharacter, the character is sent to the decoder at the bit ratespecified.

[0263] The setting of the decoder baud rate selection dip switches isdone by considering the setting of the encoder. The decoder baud ratemust be an equal or higher baud rate relative to the encoder. Forexample, it is possible to set the decoder ancillary baud rate to 9,6baud. In this case, the encoder baud rate may be set to any value from 3to 9,6 but not 19,2. If the decoder baud rate is set to a higher ratethan the encoder, the data will burst out at the decoder's baud rate.The maximum sustained baud rate is therefore controlled by the encoder.

[0264] The compression technique for the transmission of ancillary datais as follows: the encoder looks, during each 24 millisecond frameinterval, to set if any ancillary data is in its input buffer. If thereare characters in the encoder's input buffer, then the maximum number ofcharacters consistent with the selected baud rate are sent. During a 24millisecond period, the table below shows the maximum number ofcharacters per frame (at 48 kHz sampling rate) sent for each baud rate.BIT RATE NUMBER OF CHARACTERS 3 1 12 3 24 6 36 9 48 12 72 18 96 24 19247

[0265] The CODEC 1 provides no error detection or correction for theancillary data. The user assumes the responsibility for the errorcontrol strategy of this data. For example, at an error rate of le-5(which is relatively high) and an ancillary data rate of 12 baud, 1 outof every 3 characters will be received in error. Standard computer datacommunication protocol techniques might be used to maintain dataintegrity. When designing an error protection strategy, it must beremembered that the CODEC 1 may occasionally repeat the last 24milliseconds of audio under certain error conditions. The effect onaudio is nearly imperceptible. However, the ancillary data is notrepeated.

[0266] The format of the ancillary data is user defined. The presentinvention utilizes two formats for the ancillary data. The first formattreats the entire data stream as one logical (and physical) stream ofdata. The second format allows for multiplexing of various logical anddiverse data streams into one physical data stream. For example, switchclosure, RS232 , and time code data are all multiplexed into a singlephysical data stream and placed in the ancillary data stream of the ISOMPEG packet.

[0267]FIG. 1 shows a high level diagram of the asynchronous multiplexer(MUX) 3 in relation to the other CODEC components. FIG. 3 shows anisolated diagram of the multiplexer 3 in relation to encoder 1. The datarate for the multiplexer is set by software command (via remote controlconnections or keypad entry). A software command also controls a switch34 (FIG. 1) which routes the ancillary data through multiplexer 3.Multiplexer output line 36 routes the multiplexed data into the encoderinput line 38. Alternatively, if the switch 34 is in the other position,ancillary data will be routed directly to the encoder input line 38 viathe input line 32 without multiplexing. When the multiplexer 3 is used,FIG. 1 shows signals from input sources such as RS485 (line 31), RS232(line 33), contact closures (line 35), time codes (line 37), andancillary data— RS232 (line 39). FIG. 3 shows similar inputs intomultiplexer 3. These ancillary inputs are used as follows:

[0268] The RS232 I/O connector is used to provide an additional portinto the data multiplexer. It might be thought of as a second RS232ancillary port. The RS485 I/O connector is used to provide an additionaltype of port into the data multiplexer. It is a dedicated RS485 port andmight be used to control RS485 equipment.

[0269] Contact closure inputs 3 allow simple ON/OFF switches to beinterfaced into the CODEC 1. The contact closure inputs 3 areelectrically isolated from the internal circuitry by optical isolators.A plurality of optical isolated I/O lines and/or contact closure linesmight be used. Additionally, the time code inputs allow transmission oftimecode at rates of 24, 25, 29, and 3 frames per second.

[0270] Referring again to FIG. 3, the Ancillary Data Multiplexer 3multiplexes the various inputs into a composite ancillary data streamfor routing to encoder input line 38. The encoder 1 then processes thedigital audio signals (e.g. converted left and right analog inputs.AES/EBU, SPDIF, or optical) and the ancillary data stream (e.g.multiplexed composite or direct) into a compressed audio bit stream. InFIG. 3, an ISO/MPEG encoder 1 is shown, with the digital audio left andright signals, as well as a composite ancillary data stream, beingprocessed by the ISO/MPEG encoder 1 into a resulting ISO/MPEG audio bitstream Other compression techniques besides ISO/MPEG could similarly beillustrated.

[0271] Conversely, a block diagram is shown in FIG. 4 wherein theISO/MPEG Audio Bit Stream enters an ISO MPEG Decoder 2 on line 22. Thebit stream is decoded (decompressed) and the ancillary data is separatedfrom the audio data. The composite ancillary data stream enters theAncillary Data De-Multiplexer 6 through line 23. The Ancillary data isde-multiplexed into its component parts of Ancillary, RS232 , RS485 ,Time Code, and Relay Contact data, as shown by lines 61, 63, 65, 67, and69. The audio data (left and right) is output on lines 26 and 28. Asoftware command also controls a switch 64 (FIG. 1) that might route theancillary data out of decoder 2, through the de-Multiplexer 6, throughline 66, and out to ancillary data line 69. Alternatively, the ancillarydata might be routed directly from decoder output line 23, though line62, and out line 69—without multiplexing.

[0272] Referring again to FIG. 2, this more detailed block diagram ofCODEC 1 shows external ancillary data entering the ancillary data switch16 via line 39 and exiting switch 16 via line 69. (See lines 39, 69 andswitches 34, 64 in FIG. 1). Switch 16 interacts with Ancillary Data UART(Universal Asynchronous Receiver Transmitter) via connections 164 and165. Switch 16 also interacts with DSP Ancillary Data UART 169 viaconnections 166 and 167. The resulting data is sent through Switch 16 toencoder 1 via connection 162. Decoded ancillary data is sent throughSwitch 16 from decoder 2 via connection 163. Switch 16, Ancillary DataUART 168, and DSP Ancillary Data UART are controlled by Control Micro 5via control network line 155.

[0273]FIG. 2 also details the following ancillary data connections:External RS232 data is shown entering RS232 UART 17 via line 33 andexiting UART 17 via line 69.

[0274] External Time Code Data is shown entering SMPTE Time CodeInterface 172 via line 37 and exiting via line 67. Time Code Data issubsequently shown interacting with Time Code UART 174 via lines 173,175. External RS485 data is shown entering RS485 UART 176 via line 31and exiting via line 61. External optical inputs are shown enteringControl micro network 155 via line 35. Relay outputs are, shown exitingControl micro network 155 via line 65. UARTS 17, 174, 176, and Time CodeInterface 172 are controlled by Control Micro 5 via control network line155.

[0275] Ancillary data can prove to be extremely valuable because itallows the CODEC user to transmit control and message information to andfrom RS232 and RS485 equipment, on either end of the transmissionchannel, via the same compressed digital bit stream as used by the audiosignal component. The user might also send time code information andfacilitate the control of relay contacts. More importantly, the use ofancillary data does not adversely affect the ability to transmit asufficiently large amount of primary audio data.

[0276] Synchronous Ancillary Data

[0277] Referring again to FIG. 1, the CODEC 1 also provides asynchronous ancillary input data line 18 and output data line 25. Thesynchronous connections might exist separately (as shown in FIGS. 1 and2) or as part of a multi-functional input line (e.g. optical isolatedI/O, relay I/O and synchronous ancillary data I/O share a commonline—not shown). This data port is an RS232 interface, and might alsoinclude RS422 and/or RS485 capabilities.

[0278] Digital Interface Modules and Loopback Control

[0279] Referring again to FIG. 1, encoder 2 outputs a compressed audiobit stream through line 4 (and possibly more lines) into at least oneDIM 7. These modules might include, for example, the types X0.21/RS422 ,V0.35, and/or TA. These modules output the digital signals for useand/or transmission by equipment external to the CODEC. Similarly, DIM 8is connected to decoder 2 through line 81. DIM 8, using similar typemodules as DIM 7, collects the external digital signals for transmissionto decoder 2. Referring again to FIG. 2, this more detailed blockdiagram of CODEC 1 shows the compressed bit stream entering H0.221 DSP19 via line 191. DSP processes the bit stream and transfers the data,via line 192, to at least one DIM (Module types shown as 198). DIM 198interacts with TA Control UART 193 via lines 194, 195, and with DecoderDSP 2 via line 197. DIM 192 then outputs external data via line 71 andinputs external data via line 81. As discussed above, this external datais then used by external equipment such as transmitters and receivers.

[0280] Before any connection is made to the outside world, the DIMs inCODEC 1 must be defined. If the DIMs are rearranged, then the CODEC Mustbe notified via remote control software command (through the keypad orremote control interface). For DIMs that dial outside networks, twomethods of dialing exist. They are single line dialing and multiple linedialing (speed dialing).

[0281] For either mode of dialing it is possible to enable automaticreconnect. This feature allows the automatic reconnection of a droppedline. If auto reconnect is enabled when a line is dialed, then it willbe reconnected if either the far end disconnected the call, or thenetwork drops the call. If the calling end drops the call, the line willnot be automatically reconnected. This feature also allows the DIM toautomatically dial an ISDN network if, for instance, a satelliteconnection is lost.

[0282] The CODEC 1 provides for two types of loopback through loopbackcontrol module 9. Loopback is an important feature for CODEC testingpurposes. The first type is a system loopback and the second is adigital interface loopback. The system loopback is an internal loopbackwhich loops back all the digital interfaces and is set by one softwarecommand. The second type of loopback allows the user to selectindividual digital interface modules for loopback. Loopback controlmight -also be used to cause the internal CODEC clock to supply thedigital data clocks.

[0283] Satellite Receiver Interfaced with CODEC

[0284] Referring to FIG. 13, another embodiment of the disclosedinvention allows for the transmission of other information besidesaudio, including, video, text, and graphics. In this embodiment, thedigital line inputs 41 are preferably replaced with a satellite antenna46. The digital interface module 42 (or satellite receiver module)receives digital signals that are transmitted to it by the satelliteantenna 46. The digital signals, which are streams of data bits, arethen transferred to a decoder 42. The decoder decompresses the bits,whether they are audio, video, text, or graphic, and directs them to theappropriate output.

[0285] Preferably, the digital interface module 42 has the ability tostore digital information. In this alternate embodiment, the digitalinterface module (satellite receiver module) is preferably a receivercalled a “daX”. Such a receiver is available commercially under the name“daX” from Virtual Express Communications in Reno, Nev. In thisembodiment, the decoder preferably would have the capability todecompress or decode other types of compressed information such asvideo, text, and graphics. This could be facilitated by downloading therequired compression techniques into the CODEC 1 as described above.

[0286] In its operation, the satellite antenna 46 might receive digitalinformation from various sources including a remote CODEC or a remotedaX (not shown), and transfer the information to the daX receiver 42.The daX DIM 44 might also act as a switching mechanism to route thedigital bit streams to different places. It might direct informationreceived from the satellite directly to the decoder, via line 4, fordecompression and immediate output. The received, data from thesatellite receiver 42 might alternatively be directed through the daXDIM 44 to the daX 45 via line 43 for storage and later retrieval. Thedigital interface module 44 might then direct these stored data bitsfrom the daX 45 to the decoder 42 via path 4 for decoding and subsequentoutput. This embodiment also preferably allows for simultaneous storageof digital information in the DAX via path 43 and for immediate decodingof digital information via line 4 through the decoder 42.

[0287] While few preferred embodiments of the invention have beendescribed hereinabove, those of ordinary skill in the art will recognizethat these embodiments may be modified and altered without departingfrom the central spirit and scope of the invention. Thus, theembodiments described hereinabove are to be considered in all respectsas illustrative and not restrictive, the scope of the invention beingindicated by the appended claims, rather than by the foregoingdescriptions, and all changes which come within the meaning and range ofequivalency of the claims are intended to be embraced herein.

What is claimed is:
 1. An audio CODEC for providing high quality digitalaudio comprising: an analog to digital converter for converting ananalog audio signal to a digital audio bit stream; an encoder forcompressing said digital audio bit stream; a decoder for decompressingsaid compressed digital audio bit stream; an output allowing a user tomonitor the digital audio output; and at least one control for allowingsaid user to adjust said digital audio output.
 2. A method for providinghigh quality digital audio comprising the steps of: providing an inputanalog audio signal; providing at least one psycho-acoustic parameters;converting said input analog audio signal into a digital signal; codingsaid digital signal in accordance with said at least one psycho-acousticparameter; decompressing said digital signal to provide an output audiosignal; and providing an adjustment means for allowing the user toadjust said at least one psycho-acoustic parameter.
 3. The method ofclaim 2 further comprising the step of transmitting said digital signalthrough a transmission channel.